This invention relates to telephone systems, and more particularly to Digital-Subscriber Lines (DSL) carrying both voice and data traffic.
The telephone system was originally constructed for carrying voice calls. With the widespread acceptance of personal computers and wide-area data networks such as the Internet, telephone networks are carrying more and more data traffic. Digital telephone lines such as Integrated Services Digital Network (ISDN) and T1 lines carry both data and voice traffic. Some higher-speed Digital-Subscriber Lines (DSL) also reserve some bandwidth for voice calls.
FIG. 1 shows an ISDN line used for both voice and data traffic. The customer premises equipment (CPE) includes ISDN terminal adapter or modem 12 that receives a data stream from a computer. ISDN modem 12 may include one or two plain-old-telephone-service (POTS) voice ports that can be connected to standard telephone or fax equipment. ISDN modem 12 transmits data or digitized voice over telephone line 20, which is an ISDN line using digital rather than analog signaling.
At the Phone Company""s central office (CO), ISDN line card 14 terminates ISDN telephone line 20. The data or digitized voice is sent over the public switched-telephone network (PSTN) that includes circuit switched network 22. When ISDN modem 12 transmits a voice call, the call is sent over circuit switched network 22 to other voice telephones over voice lines 26, 27. However, when data is sent by ISDN modem 12, the remote number called is connected to another computer (not shown) at Internet Service Provider (ISP) 21.
ISDN telephone line 20 carries three independent channels: a low-bandwidth data channel D is primarily used for control signals, while bearer channels B1, B2 are each 64 Kbps channels, each capable of carrying one voice call. When no voice calls are being made, ISDN modem 12 routes the data stream over both bearer channels B1, B2, providing a combined bandwidth of 128 Kbps. These two bearer channels remain as two separate calls within the PSTN, since ISDN line card 14 makes two connections 24, 25 to circuit switched network 22. These two calls are sent over two lines 28, 29 to ISP 21. At ISP 21, upper-layer software 18 combines data from the two lines 28, 29 into a single data stream. Thus the single data stream from the user is split into two separate calls to ISP 21.
When the user makes or receives a voice call, one of the bearer channels is dropped and no longer carries data. The data bandwidth then falls to 64 Kbps since only one bearer channels B1 in ISDN telephone line 20 is used for data, and only one connection 24 and only one of two lines 28, 29 are used. The other bearer channel B2 is used for the voice call, connecting the user""s telephone or fax with a remote telephone or fax on voice line 26. ISDN line card 14 uses the other connection 25 to connect with circuit switched network 22 and voice line 26. When two voice calls are simultaneously made, then both bearer channels are used and no data can be transmitted. Thus ISDN modem 12 is able to drop one of the bearer channels for a voice call. Once the voice call ends, ISDN modem 12 reconnects the second bearer channel to ISP 21.
While ISDN bearer channels can be dropped and reconnected as voice and data loads change, the low bandwidth of ISDN limits its future usage. Having to separate data connections through the PSTN that must be recombined by software at the ISP is also undesirable.
FIG. 2 highlights a high-speed T1 phone line that has fixed allocations of its bandwidth to voice and data traffic. Larger corporations often connect to the telephone network over 1.5-Mps T1 or 45-Mbps T3 lines.
T1 lines can be partitioned into a number of 64 Kbit DS0 channels to carry several separate voice calls, with the remaining bandwidth used for data traffic. For example, the T1 line can have 256 Kbps allocated for voice calls, with the remaining 1.25 Mbps for data traffic. The 256 Kbps allocated for voice can carry four separate voice calls at one time.
While it is useful to allocate some of the bandwidth of a T1 line to voice calls, the allocation does not vary over time. Once the T1 line is configured for four voice calls, the allocation cannot be changed when five or more voice calls are received. The fifth caller hears a busy signal. Since calls often occur together at peak times such as just after lunch, bandwidth must be reserved for these peak times. Data bandwidth is restricted for all times of the day and night by the peak voice bandwidth than occurs only infrequently.
FIG. 3 is a graph of DSL frequency bands with a lower POTS band. FIG. 3 shows frequency bands for asymmetric DSL, or ADSL (T1.413) service using frequency-division duplex and voice calls. FIGS. 3 is not drawn with a linear scale. Plain-old-telephone service (POTS) voice calls are transmitted over low-frequency POTS band 2, as they are for standard telephone lines. POTS band 2 operates from near D.C. to 4 kHz. Since this is the same frequency range as standard telephones, ordinary telephone equipment or voice-band modems can be used over POTS band 2.
ADSL upstream channel 4 is for uploads from the customer, or for sending commands and user input from the customer to the central office side. Some embodiments may use a bi-directional channel in place of upstream channel 4. Upstream channel 4 operates at up to 138 kHz, with the data rate up to 1 Mbps.
Wide-band 5 carries the bulk of the ADSL-line bandwidth. Wide-band 5 carries ADSL data downstream to the customer at up to 8 Mbps. Wide-band 5 is a frequency band typically from 140-200 kHz up to about 1.1 MHz. The lowest frequencies are reserved for POTS. Other kinds of DSL use different frequency bands, but all use relatively high frequency bands.
While ADSL can be configured to reserve some bandwidth for services such as ISDN basic rate, POTS band 2 is frequency-limited and can carry just one voice call. Other proposed DSL services do not have a POTS band at all, and provide transport without distinguishing between voice and data services.
Special equipment is needed at both the customer premises and at the phone company""s central office where the customer""s copper phone line ends. Analog devices called frequency splitters are typically used to separate the low-frequency POTS band from the high-speed data bands. FIG. 4 is a diagram of a DSL phone line highlighting the frequency splitters.
Copper telephone line 20 is a pair of copper wires running from central office 8 to the customer. The phone customer has installed customer premises equipment 6. Since DSL uses high frequencies for data traffic and POTS uses low frequencies for voice calls, the signal received over POTS telephone line 20 must be split into high- and low frequency components. Splitter 46 contains a low-pass filter that outputs the low-frequency components from copper telephone line 20. These low-frequency components carry the voice calls that are sent to telephone set 10. Telephone set 10 is a standard POTS analog telephone set. Additional phone sets, fax machines, or voice-band modem equipment can be connected to telephone set 10 as phone-line extensions as is well-known.
Splitter 46 also contains a high-pass filter that outputs the high-frequency components to DSL modem 48. DSL modem 48 receives the high-frequency analog signal from splitter 46 and converts it to downstream digital data during the receiving window. During the transmitting window, it converts the upstream data into high-frequency analog signal. Splitter 46 mixes this high-frequency analog signal from DSL modem 48 with the low-frequency voice from telephone set 10 and transmits the combined signal over copper telephone line 20 to central office 8.
Central office 8 receives copper telephone line 20 and splits off the high-frequency components with splitter 16. The high-frequency components from splitter 16 are sent to DSL modem 47, which converts the analog high-frequency signal to an upstream digital data. DSL line card 50 includes DSL modem 47 and, in some embodiments, splitter 16. The data stream can then be connected to a high-speed data highway or backbone.
Splitter 16 sends low-frequency components to conventional telephone switch 19, which includes a line card similar to conventional line cards that terminate POTS lines. Conventional telephone switch 19 uses a PCM highway or circuit switched network that connects this voice call to remote voice-band equipment such as a telephone set.
Incoming voice calls received by conventional telephone switch 19 are combined by splitter 16 with high-frequency data traffic from DSL modem 47. The combined signal is transmitted over copper phone line 20 to customer premises equipment 6.
While such ADSL equipment is useful, only a small portion of the bandwidth is reserved for voice calls. While data rates are high, the ADSL line is not a true replacement for a T1 line, since the ADSL line cannot be partially allocated for multiple voice calls.
What is desired is a DSL system that can carry multiple voice calls. It is desired to use a high-speed DSL line to carry many voice calls. It is further desired to use idle voice bandwidth for data traffic. It is desired to flexibly allocate the bandwidth of a DSL line to voice and data traffic. It is desired to change the allocation of voice bandwidth as voice calls are received and terminated. Dynamic allocation of the bandwidth of a DSL line among voice and data traffic is desirable, without corruption or interruption of the data.
A dynamically-allocating Digital-Subscriber Line (DSL) modem dynamically allocates bandwidth among voice calls and unchannelized user data. The modem has a plurality of local voice lines. Each of the local voice lines is for carrying a voice call.
A data stream sends user data to a telephone network. A DSL connection to a DSL telephone line connects to a central office connected to the telephone network. A formatter is coupled to the DSL connection and receives the user data from the data stream and voice calls from the plurality of voice lines. It formats the user data and the voice calls into timeslots for transmitting over the DSL telephone line to the central office.
A current-format storage is coupled to control the formatter. It stores a current allocation of the timeslots. The current allocation indicates which timeslots are carrying voice calls and which timeslots are carrying the user data.
A next-format generator is coupled to the current-format storage. It generates a next allocation of the timeslots. The next allocation of the timeslots has more timeslots allocated to voice calls when a new voice call is initiated, but fewer timeslots allocated to voice calls when a voice call is terminated. Thus allocation of the timeslots for voice calls and user data is dynamically adjusted as voice calls are initiated and terminated.
In further aspects of the invention the next-format generator has an off-hook detector that is coupled to the plurality of local voice lines. It determines when a local voice line is off-hook. State means stores a current state of local voice lines that are off hook. An allocater is coupled to the state means. It changes an allocation of a timeslot from the user data to a voice call when an additional voice line is off-hook, but changes allocation of a timeslot from a voice call to the user data when a voice line is no longer off-hook.
In still further aspects of the invention the next allocation of the timeslots is transmitted to the central office over the DSL telephone line before the next allocation becomes the current allocation. Thus timeslot allocations are transmitted over the DSL telephone line before taking effect. The next allocation is latched into the current-format storage from the next-format generator when a next superframe begins to be transmitted over the DSL telephone line. Thus allocations are changed at superframe boundaries.
In other aspects the timeslots each transmit 64-Kbits per second over the DSL telephone line, and each timeslot is able to carry exactly one voice call. The DSL telephone line is divided into enough timeslots so that a maximum number of voice calls that can be simultaneously carried is at least three simultaneous voice calls.
In other aspects of the invention the superframe contains a plurality of network frames. Each network frame is synchronized to a telephone-network timing reference. Each network frame contains a plurality of low-level frames. Each low-level frame contains the timeslots that are each allocated to either voice calls or the user data. The timeslots are repeated for each low-level frame in the network frame, and the network frame is repeated for each network frame in the superframe. Thus the timeslots are repeated at a lowest level of a three-level framing structure, but allocations of the timeslots are changed only at a highest level of a framing structure.
In further aspects, each timeslot contains 8 bits, and the low-level frame is repeated at a rate of 8 kHz. Thus each timeslot carries 64 Kbits per second of data or voice-call information.